chromium/third_party/webrtc/modules/rtp_rtcp/source/packet_loss_stats.cc

/*
 *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#include "modules/rtp_rtcp/source/packet_loss_stats.h"

#include <cstdint>
#include <iterator>
#include <vector>

#include "rtc_base/checks.h"

// After this many packets are added, adding additional packets will cause the
// oldest packets to be pruned from the buffer.
static const int kBufferSize = …;

namespace webrtc {

PacketLossStats::PacketLossStats()
    : … { … }

PacketLossStats::~PacketLossStats() = default;

void PacketLossStats::AddLostPacket(uint16_t sequence_number) { … }

int PacketLossStats::GetSingleLossCount() const { … }

int PacketLossStats::GetMultipleLossEventCount() const { … }

int PacketLossStats::GetMultipleLossPacketCount() const { … }

void PacketLossStats::ComputeLossCounts(
    int* out_single_loss_count,
    int* out_multiple_loss_event_count,
    int* out_multiple_loss_packet_count) const { … }

void PacketLossStats::PruneBuffer() { … }

}  // namespace webrtc